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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
77938e640907ee7e0e63be0e8b86e5732dcb43bb
/
call
/
audio_send_stream.h
77938e6
Simulcast work to enable RID mux.
by Amit Hilbuch
· 6 years ago
c69a56e
Remove more unneeded things from ChannelSend
by Fredrik Solenberg
· 6 years ago
5571812
Adding rtcp report interval into RTCConfiguration.
by Jiawei Ou
· 6 years ago
9190b82
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
by Johannes Kron
· 6 years ago
7d76a31
Use MediaTransportInterface, for audio streams.
by Niels Möller
· 6 years ago
359d60a
Adds target rate to audio send stream stats.
by Sebastian Jansson
· 6 years ago
648d28a
Media engine and channel support for per-channel dscp values, specified by RtpParameter
by Tim Haloun
· 6 years ago
bfb444c
Adds new CryptoOption crypto_options.frame.require_frame_encryption.
by Benjamin Wright
· 6 years ago
84583f6
Enable End-to-End Encrypted Audio Payloads.
by Benjamin Wright
· 6 years ago
a12c42a
Delete root header file typedef.h.
by Niels Möller
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
b9b146c
Replace rtc::Optional with absl::optional in audio, call and video
by Danil Chapovalov
· 6 years ago
bb50ce5
Wire up MID send value to the PeerConnection API
by Steve Anton
· 6 years ago
77490b9
Pass a real audio codec pair ID to encoders that we create
by Karl Wiberg
· 6 years ago
a8b7c7f
Move remaining traces of VoiceEngine
by Fredrik Solenberg
· 7 years ago
8f5787a
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
by Fredrik Solenberg
· 7 years ago
24722b3
Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator."
by Seth Hampson
· 7 years ago
8b77aea
Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator."
by Lu Liu
· 7 years ago
d2b912a
Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
by Seth Hampson
· 7 years ago
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago
56d4609
Use the new AudioProcessing statistics everywhere.
by Ivo Creusen
· 7 years ago
b3944f0
Media track ID visibility at BWE level
by Alex Narest
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/call/audio_send_stream.h]
e1198e0
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.
by ivoc
· 7 years ago
84f6a3f
Move optional.h to webrtc/api/
by kwiberg
· 7 years ago
1acbd68
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago
abbc430
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
by eladalon
· 7 years ago
e76bd3a
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
by zstein
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
eb1fde4
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
by ossu
· 7 years ago
20a4b3f
Injectable audio encoders: WebRtcVoiceEngine and company
by ossu
· 7 years ago
4e477a1
Added a new echo likelihood stat that reports the maximum value from a previous time period.
by ivoc
· 8 years ago
f515ab8
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago
[Renamed (96%) from webrtc/api/call/audio_send_stream.h]
a8eb756
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
by aleloi
· 8 years ago
1acfbd2
Expose RtpCodecParameters to VoiceMediaInfo stats.
by hbos
· 8 years ago
ffbbcac
Support multiple timestamp rates for sending DTMF.
by solenberg
· 8 years ago
10cbb46
Fixing config for Audio BWE.
by minyue
· 8 years ago
6b825df
Using AudioOption to enable audio network adaptor.
by minyue
· 8 years ago
940b6d6
Clean up logging in AudioSendStream::SetupSendCodec().
by solenberg
· 8 years ago
189f9b1
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
by terelius
· 8 years ago
1836fd6
Clean up logging in AudioSendStream::SetupSendCodec().
by solenberg
· 8 years ago
8c63a82
Add a placeholder stat for logging the estimated residual echo likelihood.
by ivoc
· 8 years ago
7a97344
Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.
by minyue
· 8 years ago
a69d973
Move webrtc/audio_*.h to webrtc/api/call
by kjellander
· 8 years ago
[Renamed (96%) from webrtc/audio_send_stream.h]
86cc6ff
Variable audio bitrate.
by mflodman
· 8 years ago
9421853
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
by solenberg
· 8 years ago
971cab0
Configure VoE NACK through AudioSendStream::Config, for send streams.
by solenberg
· 8 years ago
6806136
Remove RED support from WebRtcVoiceEngine/MediaChannel
by kwiberg
· 8 years ago
fd8be34
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 8 years ago
6ab3db2
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
by kwiberg
· 8 years ago
65fc62e
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 8 years ago
1ba8d39
Remove webrtc/stream.h and unutilized inheritance.
by pbos
· 8 years ago
bfefb03
Replace scoped_ptr with unique_ptr everywhere
by kwiberg
· 8 years ago
8842c3e
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
3ecb5c8
Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ )
by solenberg
· 9 years ago
8886c81
- Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
by solenberg
· 9 years ago
b86d4e4
Prepare the AudioSendStream to be hooked up to send-side BWE.
by Stefan Holmer
· 9 years ago
b572768
- Remove calls to VoEDtmf from WVoE/MC.
by Fredrik Solenberg
· 9 years ago
a4527c8
Add comments about the Audio parts of the public Call API being WIP.
by Fredrik Solenberg
· 9 years ago
3a94154
Move some send stream configuration into webrtc::AudioSendStream.
by solenberg
· 9 years ago
85a0496
Implement AudioSendStream::GetStats().
by solenberg
· 9 years ago
c7a8b08
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
by solenberg
· 9 years ago
cf18b34
Align new VoE API with design.
by solenberg
· 9 years ago
2d56668
Unify Transport and newapi::Transport interfaces.
by pbos
· 9 years ago
4fbae2b
Add send transports to individual webrtc::Call streams.
by solenberg
· 9 years ago
cd67022
Define Stream base classes
by Jelena Marusic
· 9 years ago
04f4931
VoE2 API draft
by Fredrik Solenberg
· 9 years ago